Integration of WebRTC with Asterisk

Technology Limitations:

If you register with an extension, this allows for 8 to 10 simultaneous calls (depending upon license type).

For some customers, this is fine, but it really limits the size of the Contact Center volumes.

Technology Workarounds:

Within WildIX, we can register as a SIP trunk the same way you register a SIP extension.

Here is what it looks like within WilidIX:

Advanced Tab:

As an example of the successful registration on a site in the U.K.

To allow outbound calls on a trunk you will need to set up a dial plan to be successful in the calls leaving the system. This applies to both WebRTC outbound calls and Queue Call Backs. But, once setup, this isn’t a problem.

SIP Header Routing:

A big benefit of trunk registration is that you can do pretty much whatever you want to the inbound and outbound traffic.

WildIX allows us to add arbitrary SIP Headers, so as an example, I’m able to route directly to the Sale or Support skills by adding a custom header, X-XIMA-SKILL. Here is a Wildix dialplan entry for this:

CCaaS Side of things:

Outbound Dial Plan Example

NOTE: Any number starting with a + or longer than 2 digits will be dialed as if it is a user dialing it.

Inbound Dial Plan Example: (Using a DID)

This will call the Xima trunk and forward the called number in the To header (+441234567891 in this case).

It’s also adding a custom header, X-XIMA-SKILL and setting the value to “sales”

We could add many customer headers here and also manipulate the “To” header very easily.

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